The present invention relates to data streaming in packetized networks, specifically to a system and method for compensating for delay variations for continuous time signals transported over such networks.
A packet is a unit of data that is routed between an origin and a destination on a packet-switched network, such as the Internet. In a system that uses packets, data to be transmitted is separated into packets having a predefined size. The packets are assigned an identification number and a destination address, and are transmitted over the network. Once the packets arrive at the destination, their data portions are reassembled to recreate the originally-transmitted data. However, due to a variety of end-to-end delays (such as alternate data paths, for example), the packets do not necessarily arrive in the order in which they were transmitted. Thus, the packets cannot simply be reassembled as they are received. A solution to this problem is to buffer the packets as they are received. The larger the buffer, the more leeway a packet has in arriving at its destination on time. Once the packets are received, they are then reassembled in the appropriate order in accordance with their identification number.
In applications where continuous time signals are packetized, the packets are typically buffered at the receiving site and their play-out is delayed in order to compensate for the variations of the network end-to-end delays. As previously described, the buffer introduces an additional delay that allows the system to hold packets scheduled to be played-out later in time. Thus, it offers a time window over which the network end-to-end delay can vary. In the case of a non real-time application, such as audio or video streaming, the selected delay introduced by the buffer is, by design, typically set to a very large size. Such a large size minimizes the probability of receiving late packets.
However, in the case of a real-time application like video conferencing or audio conversation, large delays impair the usability of the system. Large delays in real-time applications are contrary to concept of “real-time”, wherein information is effectively delivered immediately. Thus it is preferable that the delay introduced by the buffer is minimized. However, having a smaller buffer increases the risk of losing packets if the delay of the packet arrival is greater than that provided by the buffer. Therefore, it has become an art to select a delay such that the probability of a late packet arrival is low enough that it is acceptable.
In present solutions to this problem, the size of the buffer is adjusted in accordance with arrival rate of the packets. For example, if the selected delay is observed to be too small, the buffer is increased to minimize the number of late packets. Conversely, if the selected delay is observed to be too large, the buffer is decreased to make the system appear more transparent. However, a reduction in the delay can create excess packets. Similarly, an increase in the delay can create gaps in the play-out.
Two approaches are commonly used to set the delay referred to above. Either the delay is set once for the whole session, or it is adjusted dynamically between talkspurts. A talkspurt is generally defined as a collection of packets whose data contains a continuous portion of a sound signal. In both cases, everything is done to avoid adjusting the delay during a non-silence period. It has been shown that adjusting the delay on a per-packet basis, that is, within the talkspurt, introduces gaps and slips that are damaging to the quality of the audio.
Therefore, there is a need for a system and method for reducing the excess discarding of packets and minimizing gaps when the play-out delay is adjusted. It is an object of the present invention to obviate or mitigate at least some of the above-mentioned disadvantages.